75a37b57a5
* qatar/master: APIchanges: fill in date and commit for request_sample_fmt Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders. Add support for request_sample_format in ffmpeg and ffplay. Add APIchanges entry for request_sample_fmt. Add request_sample_fmt field to AVCodecContext. Add float_interleave() to FmtConvertContext with x86-optimized versions. Remove unused make variable SEEK_REFFILE fate: remove redundant aref and vref references fate: remove do_ffmpeg_nocheck function fate: do not collect -benchmark output mpegaudiodec: remove decode_end() function fate: run aref and vref as regular tests mpegaudio: sanitise compute_antialias_* names mpeg12: add slice-threading checks to slice-threading initializers. h264: copy pixel_shift between slice threading contexts. mdec: enable frame-level multithreading. mdec.c: fix overread. Conflicts: libavcodec/aacdec.c libavcodec/ac3dec.c libavcodec/avcodec.h libavcodec/dca.c libavcodec/h264.c libavcodec/mdec.c libavcodec/mpeg12.c libavcodec/options.c libavcodec/version.h libavcodec/vorbisdec.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
2542 lines
89 KiB
C
2542 lines
89 KiB
C
/*
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* AAC decoder
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* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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*
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* AAC LATM decoder
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* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
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* Copyright (c) 2010 Janne Grunau <janne-ffmpeg@jannau.net>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* AAC decoder
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* @author Oded Shimon ( ods15 ods15 dyndns org )
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* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
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*/
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/*
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* supported tools
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*
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* Support? Name
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* N (code in SoC repo) gain control
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* Y block switching
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* Y window shapes - standard
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* N window shapes - Low Delay
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* Y filterbank - standard
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* N (code in SoC repo) filterbank - Scalable Sample Rate
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* Y Temporal Noise Shaping
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* Y Long Term Prediction
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* Y intensity stereo
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* Y channel coupling
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* Y frequency domain prediction
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* Y Perceptual Noise Substitution
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* Y Mid/Side stereo
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* N Scalable Inverse AAC Quantization
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* N Frequency Selective Switch
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* N upsampling filter
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* Y quantization & coding - AAC
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* N quantization & coding - TwinVQ
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* N quantization & coding - BSAC
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* N AAC Error Resilience tools
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* N Error Resilience payload syntax
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* N Error Protection tool
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* N CELP
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* N Silence Compression
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* N HVXC
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* N HVXC 4kbits/s VR
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* N Structured Audio tools
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* N Structured Audio Sample Bank Format
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* N MIDI
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* N Harmonic and Individual Lines plus Noise
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* N Text-To-Speech Interface
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* Y Spectral Band Replication
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* Y (not in this code) Layer-1
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* Y (not in this code) Layer-2
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* Y (not in this code) Layer-3
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* N SinuSoidal Coding (Transient, Sinusoid, Noise)
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* Y Parametric Stereo
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* N Direct Stream Transfer
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*
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* Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
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* - HE AAC v2 comprises LC AAC with Spectral Band Replication and
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Parametric Stereo.
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*/
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#include "avcodec.h"
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#include "internal.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "fft.h"
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#include "fmtconvert.h"
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#include "lpc.h"
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#include "kbdwin.h"
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#include "sinewin.h"
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#include "aac.h"
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#include "aactab.h"
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#include "aacdectab.h"
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#include "cbrt_tablegen.h"
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#include "sbr.h"
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#include "aacsbr.h"
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#include "mpeg4audio.h"
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#include "aacadtsdec.h"
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#include <assert.h>
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#include <errno.h>
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#include <math.h>
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#include <string.h>
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#if ARCH_ARM
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# include "arm/aac.h"
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#endif
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union float754 {
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float f;
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uint32_t i;
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};
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static VLC vlc_scalefactors;
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static VLC vlc_spectral[11];
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static const char overread_err[] = "Input buffer exhausted before END element found\n";
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static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
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{
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// For PCE based channel configurations map the channels solely based on tags.
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if (!ac->m4ac.chan_config) {
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return ac->tag_che_map[type][elem_id];
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}
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// For indexed channel configurations map the channels solely based on position.
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switch (ac->m4ac.chan_config) {
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case 7:
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if (ac->tags_mapped == 3 && type == TYPE_CPE) {
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ac->tags_mapped++;
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return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
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}
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case 6:
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/* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
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instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
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encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
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if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
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ac->tags_mapped++;
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return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
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}
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case 5:
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if (ac->tags_mapped == 2 && type == TYPE_CPE) {
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ac->tags_mapped++;
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return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
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}
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case 4:
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if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
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ac->tags_mapped++;
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return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
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}
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case 3:
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case 2:
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if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
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ac->tags_mapped++;
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return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
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} else if (ac->m4ac.chan_config == 2) {
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return NULL;
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}
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case 1:
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if (!ac->tags_mapped && type == TYPE_SCE) {
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ac->tags_mapped++;
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return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
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}
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default:
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return NULL;
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}
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}
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/**
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* Check for the channel element in the current channel position configuration.
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* If it exists, make sure the appropriate element is allocated and map the
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* channel order to match the internal FFmpeg channel layout.
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*
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* @param che_pos current channel position configuration
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* @param type channel element type
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* @param id channel element id
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* @param channels count of the number of channels in the configuration
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static av_cold int che_configure(AACContext *ac,
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enum ChannelPosition che_pos[4][MAX_ELEM_ID],
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int type, int id, int *channels)
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{
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if (che_pos[type][id]) {
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if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
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return AVERROR(ENOMEM);
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ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
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if (type != TYPE_CCE) {
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ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
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if (type == TYPE_CPE ||
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(type == TYPE_SCE && ac->m4ac.ps == 1)) {
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ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
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}
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}
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} else {
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if (ac->che[type][id])
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ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
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av_freep(&ac->che[type][id]);
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}
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return 0;
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}
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/**
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* Configure output channel order based on the current program configuration element.
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*
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* @param che_pos current channel position configuration
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* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static av_cold int output_configure(AACContext *ac,
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enum ChannelPosition che_pos[4][MAX_ELEM_ID],
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enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
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int channel_config, enum OCStatus oc_type)
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{
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AVCodecContext *avctx = ac->avctx;
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int i, type, channels = 0, ret;
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if (new_che_pos != che_pos)
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memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
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if (channel_config) {
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for (i = 0; i < tags_per_config[channel_config]; i++) {
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if ((ret = che_configure(ac, che_pos,
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aac_channel_layout_map[channel_config - 1][i][0],
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aac_channel_layout_map[channel_config - 1][i][1],
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&channels)))
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return ret;
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}
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memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
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avctx->channel_layout = aac_channel_layout[channel_config - 1];
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} else {
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/* Allocate or free elements depending on if they are in the
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* current program configuration.
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*
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* Set up default 1:1 output mapping.
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*
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* For a 5.1 stream the output order will be:
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* [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
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*/
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for (i = 0; i < MAX_ELEM_ID; i++) {
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for (type = 0; type < 4; type++) {
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if ((ret = che_configure(ac, che_pos, type, i, &channels)))
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return ret;
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}
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}
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memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
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avctx->channel_layout = 0;
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}
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avctx->channels = channels;
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ac->output_configured = oc_type;
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return 0;
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}
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/**
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* Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
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*
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* @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
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* @param sce_map mono (Single Channel Element) map
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* @param type speaker type/position for these channels
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*/
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static void decode_channel_map(enum ChannelPosition *cpe_map,
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enum ChannelPosition *sce_map,
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enum ChannelPosition type,
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GetBitContext *gb, int n)
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{
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while (n--) {
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enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
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map[get_bits(gb, 4)] = type;
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}
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}
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/**
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* Decode program configuration element; reference: table 4.2.
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*
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* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
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enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
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GetBitContext *gb)
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{
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int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
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int comment_len;
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skip_bits(gb, 2); // object_type
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sampling_index = get_bits(gb, 4);
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if (m4ac->sampling_index != sampling_index)
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av_log(avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
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num_front = get_bits(gb, 4);
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num_side = get_bits(gb, 4);
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num_back = get_bits(gb, 4);
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num_lfe = get_bits(gb, 2);
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num_assoc_data = get_bits(gb, 3);
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num_cc = get_bits(gb, 4);
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if (get_bits1(gb))
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skip_bits(gb, 4); // mono_mixdown_tag
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if (get_bits1(gb))
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skip_bits(gb, 4); // stereo_mixdown_tag
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if (get_bits1(gb))
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skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
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if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
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av_log(avctx, AV_LOG_ERROR, overread_err);
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return -1;
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}
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decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
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decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side );
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decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back );
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decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe );
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skip_bits_long(gb, 4 * num_assoc_data);
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decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc );
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align_get_bits(gb);
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/* comment field, first byte is length */
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comment_len = get_bits(gb, 8) * 8;
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if (get_bits_left(gb) < comment_len) {
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av_log(avctx, AV_LOG_ERROR, overread_err);
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return -1;
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}
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skip_bits_long(gb, comment_len);
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return 0;
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}
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/**
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* Set up channel positions based on a default channel configuration
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* as specified in table 1.17.
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*
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* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static av_cold int set_default_channel_config(AVCodecContext *avctx,
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enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
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int channel_config)
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{
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if (channel_config < 1 || channel_config > 7) {
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av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
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channel_config);
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return -1;
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}
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/* default channel configurations:
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*
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* 1ch : front center (mono)
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* 2ch : L + R (stereo)
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* 3ch : front center + L + R
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* 4ch : front center + L + R + back center
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* 5ch : front center + L + R + back stereo
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* 6ch : front center + L + R + back stereo + LFE
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* 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
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*/
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if (channel_config != 2)
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new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
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if (channel_config > 1)
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new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
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if (channel_config == 4)
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new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center
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if (channel_config > 4)
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new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
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= AAC_CHANNEL_BACK; // back stereo
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if (channel_config > 5)
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new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE
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if (channel_config == 7)
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new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
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return 0;
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}
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/**
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* Decode GA "General Audio" specific configuration; reference: table 4.1.
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*
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* @param ac pointer to AACContext, may be null
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* @param avctx pointer to AVCCodecContext, used for logging
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
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GetBitContext *gb,
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MPEG4AudioConfig *m4ac,
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int channel_config)
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{
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enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
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int extension_flag, ret;
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if (get_bits1(gb)) { // frameLengthFlag
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av_log_missing_feature(avctx, "960/120 MDCT window is", 1);
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return -1;
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}
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if (get_bits1(gb)) // dependsOnCoreCoder
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skip_bits(gb, 14); // coreCoderDelay
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extension_flag = get_bits1(gb);
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if (m4ac->object_type == AOT_AAC_SCALABLE ||
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m4ac->object_type == AOT_ER_AAC_SCALABLE)
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skip_bits(gb, 3); // layerNr
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memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
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if (channel_config == 0) {
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skip_bits(gb, 4); // element_instance_tag
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if ((ret = decode_pce(avctx, m4ac, new_che_pos, gb)))
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return ret;
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} else {
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if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
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return ret;
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}
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if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
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return ret;
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if (extension_flag) {
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switch (m4ac->object_type) {
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case AOT_ER_BSAC:
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skip_bits(gb, 5); // numOfSubFrame
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skip_bits(gb, 11); // layer_length
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break;
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case AOT_ER_AAC_LC:
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case AOT_ER_AAC_LTP:
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case AOT_ER_AAC_SCALABLE:
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case AOT_ER_AAC_LD:
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skip_bits(gb, 3); /* aacSectionDataResilienceFlag
|
|
* aacScalefactorDataResilienceFlag
|
|
* aacSpectralDataResilienceFlag
|
|
*/
|
|
break;
|
|
}
|
|
skip_bits1(gb); // extensionFlag3 (TBD in version 3)
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode audio specific configuration; reference: table 1.13.
|
|
*
|
|
* @param ac pointer to AACContext, may be null
|
|
* @param avctx pointer to AVCCodecContext, used for logging
|
|
* @param m4ac pointer to MPEG4AudioConfig, used for parsing
|
|
* @param data pointer to AVCodecContext extradata
|
|
* @param data_size size of AVCCodecContext extradata
|
|
*
|
|
* @return Returns error status or number of consumed bits. <0 - error
|
|
*/
|
|
static int decode_audio_specific_config(AACContext *ac,
|
|
AVCodecContext *avctx,
|
|
MPEG4AudioConfig *m4ac,
|
|
const uint8_t *data, int data_size)
|
|
{
|
|
GetBitContext gb;
|
|
int i;
|
|
|
|
av_dlog(avctx, "extradata size %d\n", avctx->extradata_size);
|
|
for (i = 0; i < avctx->extradata_size; i++)
|
|
av_dlog(avctx, "%02x ", avctx->extradata[i]);
|
|
av_dlog(avctx, "\n");
|
|
|
|
init_get_bits(&gb, data, data_size * 8);
|
|
|
|
if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0)
|
|
return -1;
|
|
if (m4ac->sampling_index > 12) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
|
|
return -1;
|
|
}
|
|
if (m4ac->sbr == 1 && m4ac->ps == -1)
|
|
m4ac->ps = 1;
|
|
|
|
skip_bits_long(&gb, i);
|
|
|
|
switch (m4ac->object_type) {
|
|
case AOT_AAC_MAIN:
|
|
case AOT_AAC_LC:
|
|
case AOT_AAC_LTP:
|
|
if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
|
|
return -1;
|
|
break;
|
|
default:
|
|
av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
|
|
m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
|
|
return -1;
|
|
}
|
|
|
|
av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
|
|
m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
|
|
m4ac->sample_rate, m4ac->sbr, m4ac->ps);
|
|
|
|
return get_bits_count(&gb);
|
|
}
|
|
|
|
/**
|
|
* linear congruential pseudorandom number generator
|
|
*
|
|
* @param previous_val pointer to the current state of the generator
|
|
*
|
|
* @return Returns a 32-bit pseudorandom integer
|
|
*/
|
|
static av_always_inline int lcg_random(int previous_val)
|
|
{
|
|
return previous_val * 1664525 + 1013904223;
|
|
}
|
|
|
|
static av_always_inline void reset_predict_state(PredictorState *ps)
|
|
{
|
|
ps->r0 = 0.0f;
|
|
ps->r1 = 0.0f;
|
|
ps->cor0 = 0.0f;
|
|
ps->cor1 = 0.0f;
|
|
ps->var0 = 1.0f;
|
|
ps->var1 = 1.0f;
|
|
}
|
|
|
|
static void reset_all_predictors(PredictorState *ps)
|
|
{
|
|
int i;
|
|
for (i = 0; i < MAX_PREDICTORS; i++)
|
|
reset_predict_state(&ps[i]);
|
|
}
|
|
|
|
static void reset_predictor_group(PredictorState *ps, int group_num)
|
|
{
|
|
int i;
|
|
for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
|
|
reset_predict_state(&ps[i]);
|
|
}
|
|
|
|
#define AAC_INIT_VLC_STATIC(num, size) \
|
|
INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
|
|
ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
|
|
ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
|
|
size);
|
|
|
|
static av_cold int aac_decode_init(AVCodecContext *avctx)
|
|
{
|
|
AACContext *ac = avctx->priv_data;
|
|
float output_scale_factor;
|
|
|
|
ac->avctx = avctx;
|
|
ac->m4ac.sample_rate = avctx->sample_rate;
|
|
|
|
if (avctx->extradata_size > 0) {
|
|
if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
|
|
avctx->extradata,
|
|
avctx->extradata_size) < 0)
|
|
return -1;
|
|
}
|
|
|
|
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
|
|
output_scale_factor = 1.0 / 32768.0;
|
|
} else {
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
|
|
output_scale_factor = 1.0;
|
|
}
|
|
|
|
AAC_INIT_VLC_STATIC( 0, 304);
|
|
AAC_INIT_VLC_STATIC( 1, 270);
|
|
AAC_INIT_VLC_STATIC( 2, 550);
|
|
AAC_INIT_VLC_STATIC( 3, 300);
|
|
AAC_INIT_VLC_STATIC( 4, 328);
|
|
AAC_INIT_VLC_STATIC( 5, 294);
|
|
AAC_INIT_VLC_STATIC( 6, 306);
|
|
AAC_INIT_VLC_STATIC( 7, 268);
|
|
AAC_INIT_VLC_STATIC( 8, 510);
|
|
AAC_INIT_VLC_STATIC( 9, 366);
|
|
AAC_INIT_VLC_STATIC(10, 462);
|
|
|
|
ff_aac_sbr_init();
|
|
|
|
dsputil_init(&ac->dsp, avctx);
|
|
ff_fmt_convert_init(&ac->fmt_conv, avctx);
|
|
|
|
ac->random_state = 0x1f2e3d4c;
|
|
|
|
ff_aac_tableinit();
|
|
|
|
INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
|
|
ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
|
|
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
|
|
352);
|
|
|
|
ff_mdct_init(&ac->mdct, 11, 1, output_scale_factor/1024.0);
|
|
ff_mdct_init(&ac->mdct_small, 8, 1, output_scale_factor/128.0);
|
|
ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0/output_scale_factor);
|
|
// window initialization
|
|
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
|
|
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
|
|
ff_init_ff_sine_windows(10);
|
|
ff_init_ff_sine_windows( 7);
|
|
|
|
cbrt_tableinit();
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Skip data_stream_element; reference: table 4.10.
|
|
*/
|
|
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
|
|
{
|
|
int byte_align = get_bits1(gb);
|
|
int count = get_bits(gb, 8);
|
|
if (count == 255)
|
|
count += get_bits(gb, 8);
|
|
if (byte_align)
|
|
align_get_bits(gb);
|
|
|
|
if (get_bits_left(gb) < 8 * count) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, overread_err);
|
|
return -1;
|
|
}
|
|
skip_bits_long(gb, 8 * count);
|
|
return 0;
|
|
}
|
|
|
|
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
|
|
GetBitContext *gb)
|
|
{
|
|
int sfb;
|
|
if (get_bits1(gb)) {
|
|
ics->predictor_reset_group = get_bits(gb, 5);
|
|
if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
|
|
return -1;
|
|
}
|
|
}
|
|
for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
|
|
ics->prediction_used[sfb] = get_bits1(gb);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode Long Term Prediction data; reference: table 4.xx.
|
|
*/
|
|
static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
|
|
GetBitContext *gb, uint8_t max_sfb)
|
|
{
|
|
int sfb;
|
|
|
|
ltp->lag = get_bits(gb, 11);
|
|
ltp->coef = ltp_coef[get_bits(gb, 3)];
|
|
for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
|
|
ltp->used[sfb] = get_bits1(gb);
|
|
}
|
|
|
|
/**
|
|
* Decode Individual Channel Stream info; reference: table 4.6.
|
|
*
|
|
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
|
|
*/
|
|
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
|
|
GetBitContext *gb, int common_window)
|
|
{
|
|
if (get_bits1(gb)) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
|
|
memset(ics, 0, sizeof(IndividualChannelStream));
|
|
return -1;
|
|
}
|
|
ics->window_sequence[1] = ics->window_sequence[0];
|
|
ics->window_sequence[0] = get_bits(gb, 2);
|
|
ics->use_kb_window[1] = ics->use_kb_window[0];
|
|
ics->use_kb_window[0] = get_bits1(gb);
|
|
ics->num_window_groups = 1;
|
|
ics->group_len[0] = 1;
|
|
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
|
|
int i;
|
|
ics->max_sfb = get_bits(gb, 4);
|
|
for (i = 0; i < 7; i++) {
|
|
if (get_bits1(gb)) {
|
|
ics->group_len[ics->num_window_groups - 1]++;
|
|
} else {
|
|
ics->num_window_groups++;
|
|
ics->group_len[ics->num_window_groups - 1] = 1;
|
|
}
|
|
}
|
|
ics->num_windows = 8;
|
|
ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index];
|
|
ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index];
|
|
ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index];
|
|
ics->predictor_present = 0;
|
|
} else {
|
|
ics->max_sfb = get_bits(gb, 6);
|
|
ics->num_windows = 1;
|
|
ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index];
|
|
ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index];
|
|
ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
|
|
ics->predictor_present = get_bits1(gb);
|
|
ics->predictor_reset_group = 0;
|
|
if (ics->predictor_present) {
|
|
if (ac->m4ac.object_type == AOT_AAC_MAIN) {
|
|
if (decode_prediction(ac, ics, gb)) {
|
|
memset(ics, 0, sizeof(IndividualChannelStream));
|
|
return -1;
|
|
}
|
|
} else if (ac->m4ac.object_type == AOT_AAC_LC) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
|
|
memset(ics, 0, sizeof(IndividualChannelStream));
|
|
return -1;
|
|
} else {
|
|
if ((ics->ltp.present = get_bits(gb, 1)))
|
|
decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (ics->max_sfb > ics->num_swb) {
|
|
av_log(ac->avctx, AV_LOG_ERROR,
|
|
"Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
|
|
ics->max_sfb, ics->num_swb);
|
|
memset(ics, 0, sizeof(IndividualChannelStream));
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode band types (section_data payload); reference: table 4.46.
|
|
*
|
|
* @param band_type array of the used band type
|
|
* @param band_type_run_end array of the last scalefactor band of a band type run
|
|
*
|
|
* @return Returns error status. 0 - OK, !0 - error
|
|
*/
|
|
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
|
|
int band_type_run_end[120], GetBitContext *gb,
|
|
IndividualChannelStream *ics)
|
|
{
|
|
int g, idx = 0;
|
|
const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
|
|
for (g = 0; g < ics->num_window_groups; g++) {
|
|
int k = 0;
|
|
while (k < ics->max_sfb) {
|
|
uint8_t sect_end = k;
|
|
int sect_len_incr;
|
|
int sect_band_type = get_bits(gb, 4);
|
|
if (sect_band_type == 12) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
|
|
return -1;
|
|
}
|
|
while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
|
|
sect_end += sect_len_incr;
|
|
sect_end += sect_len_incr;
|
|
if (get_bits_left(gb) < 0) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, overread_err);
|
|
return -1;
|
|
}
|
|
if (sect_end > ics->max_sfb) {
|
|
av_log(ac->avctx, AV_LOG_ERROR,
|
|
"Number of bands (%d) exceeds limit (%d).\n",
|
|
sect_end, ics->max_sfb);
|
|
return -1;
|
|
}
|
|
for (; k < sect_end; k++) {
|
|
band_type [idx] = sect_band_type;
|
|
band_type_run_end[idx++] = sect_end;
|
|
}
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode scalefactors; reference: table 4.47.
|
|
*
|
|
* @param global_gain first scalefactor value as scalefactors are differentially coded
|
|
* @param band_type array of the used band type
|
|
* @param band_type_run_end array of the last scalefactor band of a band type run
|
|
* @param sf array of scalefactors or intensity stereo positions
|
|
*
|
|
* @return Returns error status. 0 - OK, !0 - error
|
|
*/
|
|
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
|
|
unsigned int global_gain,
|
|
IndividualChannelStream *ics,
|
|
enum BandType band_type[120],
|
|
int band_type_run_end[120])
|
|
{
|
|
int g, i, idx = 0;
|
|
int offset[3] = { global_gain, global_gain - 90, 0 };
|
|
int clipped_offset;
|
|
int noise_flag = 1;
|
|
static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
|
|
for (g = 0; g < ics->num_window_groups; g++) {
|
|
for (i = 0; i < ics->max_sfb;) {
|
|
int run_end = band_type_run_end[idx];
|
|
if (band_type[idx] == ZERO_BT) {
|
|
for (; i < run_end; i++, idx++)
|
|
sf[idx] = 0.;
|
|
} else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
|
|
for (; i < run_end; i++, idx++) {
|
|
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
|
|
clipped_offset = av_clip(offset[2], -155, 100);
|
|
if (offset[2] != clipped_offset) {
|
|
av_log_ask_for_sample(ac->avctx, "Intensity stereo "
|
|
"position clipped (%d -> %d).\nIf you heard an "
|
|
"audible artifact, there may be a bug in the "
|
|
"decoder. ", offset[2], clipped_offset);
|
|
}
|
|
sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
|
|
}
|
|
} else if (band_type[idx] == NOISE_BT) {
|
|
for (; i < run_end; i++, idx++) {
|
|
if (noise_flag-- > 0)
|
|
offset[1] += get_bits(gb, 9) - 256;
|
|
else
|
|
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
|
|
clipped_offset = av_clip(offset[1], -100, 155);
|
|
if (offset[2] != clipped_offset) {
|
|
av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
|
|
"(%d -> %d).\nIf you heard an audible "
|
|
"artifact, there may be a bug in the decoder. ",
|
|
offset[1], clipped_offset);
|
|
}
|
|
sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
|
|
}
|
|
} else {
|
|
for (; i < run_end; i++, idx++) {
|
|
offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
|
|
if (offset[0] > 255U) {
|
|
av_log(ac->avctx, AV_LOG_ERROR,
|
|
"%s (%d) out of range.\n", sf_str[0], offset[0]);
|
|
return -1;
|
|
}
|
|
sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode pulse data; reference: table 4.7.
|
|
*/
|
|
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
|
|
const uint16_t *swb_offset, int num_swb)
|
|
{
|
|
int i, pulse_swb;
|
|
pulse->num_pulse = get_bits(gb, 2) + 1;
|
|
pulse_swb = get_bits(gb, 6);
|
|
if (pulse_swb >= num_swb)
|
|
return -1;
|
|
pulse->pos[0] = swb_offset[pulse_swb];
|
|
pulse->pos[0] += get_bits(gb, 5);
|
|
if (pulse->pos[0] > 1023)
|
|
return -1;
|
|
pulse->amp[0] = get_bits(gb, 4);
|
|
for (i = 1; i < pulse->num_pulse; i++) {
|
|
pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
|
|
if (pulse->pos[i] > 1023)
|
|
return -1;
|
|
pulse->amp[i] = get_bits(gb, 4);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode Temporal Noise Shaping data; reference: table 4.48.
|
|
*
|
|
* @return Returns error status. 0 - OK, !0 - error
|
|
*/
|
|
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
|
|
GetBitContext *gb, const IndividualChannelStream *ics)
|
|
{
|
|
int w, filt, i, coef_len, coef_res, coef_compress;
|
|
const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
|
|
const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
|
|
for (w = 0; w < ics->num_windows; w++) {
|
|
if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
|
|
coef_res = get_bits1(gb);
|
|
|
|
for (filt = 0; filt < tns->n_filt[w]; filt++) {
|
|
int tmp2_idx;
|
|
tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
|
|
|
|
if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
|
|
tns->order[w][filt], tns_max_order);
|
|
tns->order[w][filt] = 0;
|
|
return -1;
|
|
}
|
|
if (tns->order[w][filt]) {
|
|
tns->direction[w][filt] = get_bits1(gb);
|
|
coef_compress = get_bits1(gb);
|
|
coef_len = coef_res + 3 - coef_compress;
|
|
tmp2_idx = 2 * coef_compress + coef_res;
|
|
|
|
for (i = 0; i < tns->order[w][filt]; i++)
|
|
tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode Mid/Side data; reference: table 4.54.
|
|
*
|
|
* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
|
|
* [1] mask is decoded from bitstream; [2] mask is all 1s;
|
|
* [3] reserved for scalable AAC
|
|
*/
|
|
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
|
|
int ms_present)
|
|
{
|
|
int idx;
|
|
if (ms_present == 1) {
|
|
for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
|
|
cpe->ms_mask[idx] = get_bits1(gb);
|
|
} else if (ms_present == 2) {
|
|
memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
|
|
}
|
|
}
|
|
|
|
#ifndef VMUL2
|
|
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
|
|
const float *scale)
|
|
{
|
|
float s = *scale;
|
|
*dst++ = v[idx & 15] * s;
|
|
*dst++ = v[idx>>4 & 15] * s;
|
|
return dst;
|
|
}
|
|
#endif
|
|
|
|
#ifndef VMUL4
|
|
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
|
|
const float *scale)
|
|
{
|
|
float s = *scale;
|
|
*dst++ = v[idx & 3] * s;
|
|
*dst++ = v[idx>>2 & 3] * s;
|
|
*dst++ = v[idx>>4 & 3] * s;
|
|
*dst++ = v[idx>>6 & 3] * s;
|
|
return dst;
|
|
}
|
|
#endif
|
|
|
|
#ifndef VMUL2S
|
|
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
|
|
unsigned sign, const float *scale)
|
|
{
|
|
union float754 s0, s1;
|
|
|
|
s0.f = s1.f = *scale;
|
|
s0.i ^= sign >> 1 << 31;
|
|
s1.i ^= sign << 31;
|
|
|
|
*dst++ = v[idx & 15] * s0.f;
|
|
*dst++ = v[idx>>4 & 15] * s1.f;
|
|
|
|
return dst;
|
|
}
|
|
#endif
|
|
|
|
#ifndef VMUL4S
|
|
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
|
|
unsigned sign, const float *scale)
|
|
{
|
|
unsigned nz = idx >> 12;
|
|
union float754 s = { .f = *scale };
|
|
union float754 t;
|
|
|
|
t.i = s.i ^ (sign & 1U<<31);
|
|
*dst++ = v[idx & 3] * t.f;
|
|
|
|
sign <<= nz & 1; nz >>= 1;
|
|
t.i = s.i ^ (sign & 1U<<31);
|
|
*dst++ = v[idx>>2 & 3] * t.f;
|
|
|
|
sign <<= nz & 1; nz >>= 1;
|
|
t.i = s.i ^ (sign & 1U<<31);
|
|
*dst++ = v[idx>>4 & 3] * t.f;
|
|
|
|
sign <<= nz & 1; nz >>= 1;
|
|
t.i = s.i ^ (sign & 1U<<31);
|
|
*dst++ = v[idx>>6 & 3] * t.f;
|
|
|
|
return dst;
|
|
}
|
|
#endif
|
|
|
|
/**
|
|
* Decode spectral data; reference: table 4.50.
|
|
* Dequantize and scale spectral data; reference: 4.6.3.3.
|
|
*
|
|
* @param coef array of dequantized, scaled spectral data
|
|
* @param sf array of scalefactors or intensity stereo positions
|
|
* @param pulse_present set if pulses are present
|
|
* @param pulse pointer to pulse data struct
|
|
* @param band_type array of the used band type
|
|
*
|
|
* @return Returns error status. 0 - OK, !0 - error
|
|
*/
|
|
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
|
|
GetBitContext *gb, const float sf[120],
|
|
int pulse_present, const Pulse *pulse,
|
|
const IndividualChannelStream *ics,
|
|
enum BandType band_type[120])
|
|
{
|
|
int i, k, g, idx = 0;
|
|
const int c = 1024 / ics->num_windows;
|
|
const uint16_t *offsets = ics->swb_offset;
|
|
float *coef_base = coef;
|
|
|
|
for (g = 0; g < ics->num_windows; g++)
|
|
memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
|
|
|
|
for (g = 0; g < ics->num_window_groups; g++) {
|
|
unsigned g_len = ics->group_len[g];
|
|
|
|
for (i = 0; i < ics->max_sfb; i++, idx++) {
|
|
const unsigned cbt_m1 = band_type[idx] - 1;
|
|
float *cfo = coef + offsets[i];
|
|
int off_len = offsets[i + 1] - offsets[i];
|
|
int group;
|
|
|
|
if (cbt_m1 >= INTENSITY_BT2 - 1) {
|
|
for (group = 0; group < g_len; group++, cfo+=128) {
|
|
memset(cfo, 0, off_len * sizeof(float));
|
|
}
|
|
} else if (cbt_m1 == NOISE_BT - 1) {
|
|
for (group = 0; group < g_len; group++, cfo+=128) {
|
|
float scale;
|
|
float band_energy;
|
|
|
|
for (k = 0; k < off_len; k++) {
|
|
ac->random_state = lcg_random(ac->random_state);
|
|
cfo[k] = ac->random_state;
|
|
}
|
|
|
|
band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
|
|
scale = sf[idx] / sqrtf(band_energy);
|
|
ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
|
|
}
|
|
} else {
|
|
const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
|
|
const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
|
|
VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
|
|
OPEN_READER(re, gb);
|
|
|
|
switch (cbt_m1 >> 1) {
|
|
case 0:
|
|
for (group = 0; group < g_len; group++, cfo+=128) {
|
|
float *cf = cfo;
|
|
int len = off_len;
|
|
|
|
do {
|
|
int code;
|
|
unsigned cb_idx;
|
|
|
|
UPDATE_CACHE(re, gb);
|
|
GET_VLC(code, re, gb, vlc_tab, 8, 2);
|
|
cb_idx = cb_vector_idx[code];
|
|
cf = VMUL4(cf, vq, cb_idx, sf + idx);
|
|
} while (len -= 4);
|
|
}
|
|
break;
|
|
|
|
case 1:
|
|
for (group = 0; group < g_len; group++, cfo+=128) {
|
|
float *cf = cfo;
|
|
int len = off_len;
|
|
|
|
do {
|
|
int code;
|
|
unsigned nnz;
|
|
unsigned cb_idx;
|
|
uint32_t bits;
|
|
|
|
UPDATE_CACHE(re, gb);
|
|
GET_VLC(code, re, gb, vlc_tab, 8, 2);
|
|
cb_idx = cb_vector_idx[code];
|
|
nnz = cb_idx >> 8 & 15;
|
|
bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
|
|
LAST_SKIP_BITS(re, gb, nnz);
|
|
cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
|
|
} while (len -= 4);
|
|
}
|
|
break;
|
|
|
|
case 2:
|
|
for (group = 0; group < g_len; group++, cfo+=128) {
|
|
float *cf = cfo;
|
|
int len = off_len;
|
|
|
|
do {
|
|
int code;
|
|
unsigned cb_idx;
|
|
|
|
UPDATE_CACHE(re, gb);
|
|
GET_VLC(code, re, gb, vlc_tab, 8, 2);
|
|
cb_idx = cb_vector_idx[code];
|
|
cf = VMUL2(cf, vq, cb_idx, sf + idx);
|
|
} while (len -= 2);
|
|
}
|
|
break;
|
|
|
|
case 3:
|
|
case 4:
|
|
for (group = 0; group < g_len; group++, cfo+=128) {
|
|
float *cf = cfo;
|
|
int len = off_len;
|
|
|
|
do {
|
|
int code;
|
|
unsigned nnz;
|
|
unsigned cb_idx;
|
|
unsigned sign;
|
|
|
|
UPDATE_CACHE(re, gb);
|
|
GET_VLC(code, re, gb, vlc_tab, 8, 2);
|
|
cb_idx = cb_vector_idx[code];
|
|
nnz = cb_idx >> 8 & 15;
|
|
sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
|
|
LAST_SKIP_BITS(re, gb, nnz);
|
|
cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
|
|
} while (len -= 2);
|
|
}
|
|
break;
|
|
|
|
default:
|
|
for (group = 0; group < g_len; group++, cfo+=128) {
|
|
float *cf = cfo;
|
|
uint32_t *icf = (uint32_t *) cf;
|
|
int len = off_len;
|
|
|
|
do {
|
|
int code;
|
|
unsigned nzt, nnz;
|
|
unsigned cb_idx;
|
|
uint32_t bits;
|
|
int j;
|
|
|
|
UPDATE_CACHE(re, gb);
|
|
GET_VLC(code, re, gb, vlc_tab, 8, 2);
|
|
|
|
if (!code) {
|
|
*icf++ = 0;
|
|
*icf++ = 0;
|
|
continue;
|
|
}
|
|
|
|
cb_idx = cb_vector_idx[code];
|
|
nnz = cb_idx >> 12;
|
|
nzt = cb_idx >> 8;
|
|
bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
|
|
LAST_SKIP_BITS(re, gb, nnz);
|
|
|
|
for (j = 0; j < 2; j++) {
|
|
if (nzt & 1<<j) {
|
|
uint32_t b;
|
|
int n;
|
|
/* The total length of escape_sequence must be < 22 bits according
|
|
to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
|
|
UPDATE_CACHE(re, gb);
|
|
b = GET_CACHE(re, gb);
|
|
b = 31 - av_log2(~b);
|
|
|
|
if (b > 8) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
|
|
return -1;
|
|
}
|
|
|
|
SKIP_BITS(re, gb, b + 1);
|
|
b += 4;
|
|
n = (1 << b) + SHOW_UBITS(re, gb, b);
|
|
LAST_SKIP_BITS(re, gb, b);
|
|
*icf++ = cbrt_tab[n] | (bits & 1U<<31);
|
|
bits <<= 1;
|
|
} else {
|
|
unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
|
|
*icf++ = (bits & 1U<<31) | v;
|
|
bits <<= !!v;
|
|
}
|
|
cb_idx >>= 4;
|
|
}
|
|
} while (len -= 2);
|
|
|
|
ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
|
|
}
|
|
}
|
|
|
|
CLOSE_READER(re, gb);
|
|
}
|
|
}
|
|
coef += g_len << 7;
|
|
}
|
|
|
|
if (pulse_present) {
|
|
idx = 0;
|
|
for (i = 0; i < pulse->num_pulse; i++) {
|
|
float co = coef_base[ pulse->pos[i] ];
|
|
while (offsets[idx + 1] <= pulse->pos[i])
|
|
idx++;
|
|
if (band_type[idx] != NOISE_BT && sf[idx]) {
|
|
float ico = -pulse->amp[i];
|
|
if (co) {
|
|
co /= sf[idx];
|
|
ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
|
|
}
|
|
coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
|
|
}
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static av_always_inline float flt16_round(float pf)
|
|
{
|
|
union float754 tmp;
|
|
tmp.f = pf;
|
|
tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
|
|
return tmp.f;
|
|
}
|
|
|
|
static av_always_inline float flt16_even(float pf)
|
|
{
|
|
union float754 tmp;
|
|
tmp.f = pf;
|
|
tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
|
|
return tmp.f;
|
|
}
|
|
|
|
static av_always_inline float flt16_trunc(float pf)
|
|
{
|
|
union float754 pun;
|
|
pun.f = pf;
|
|
pun.i &= 0xFFFF0000U;
|
|
return pun.f;
|
|
}
|
|
|
|
static av_always_inline void predict(PredictorState *ps, float *coef,
|
|
int output_enable)
|
|
{
|
|
const float a = 0.953125; // 61.0 / 64
|
|
const float alpha = 0.90625; // 29.0 / 32
|
|
float e0, e1;
|
|
float pv;
|
|
float k1, k2;
|
|
float r0 = ps->r0, r1 = ps->r1;
|
|
float cor0 = ps->cor0, cor1 = ps->cor1;
|
|
float var0 = ps->var0, var1 = ps->var1;
|
|
|
|
k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
|
|
k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
|
|
|
|
pv = flt16_round(k1 * r0 + k2 * r1);
|
|
if (output_enable)
|
|
*coef += pv;
|
|
|
|
e0 = *coef;
|
|
e1 = e0 - k1 * r0;
|
|
|
|
ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
|
|
ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
|
|
ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
|
|
ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
|
|
|
|
ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
|
|
ps->r0 = flt16_trunc(a * e0);
|
|
}
|
|
|
|
/**
|
|
* Apply AAC-Main style frequency domain prediction.
|
|
*/
|
|
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
|
|
{
|
|
int sfb, k;
|
|
|
|
if (!sce->ics.predictor_initialized) {
|
|
reset_all_predictors(sce->predictor_state);
|
|
sce->ics.predictor_initialized = 1;
|
|
}
|
|
|
|
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
|
|
for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
|
|
for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
|
|
predict(&sce->predictor_state[k], &sce->coeffs[k],
|
|
sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
|
|
}
|
|
}
|
|
if (sce->ics.predictor_reset_group)
|
|
reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
|
|
} else
|
|
reset_all_predictors(sce->predictor_state);
|
|
}
|
|
|
|
/**
|
|
* Decode an individual_channel_stream payload; reference: table 4.44.
|
|
*
|
|
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
|
|
* @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
|
|
*
|
|
* @return Returns error status. 0 - OK, !0 - error
|
|
*/
|
|
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
|
|
GetBitContext *gb, int common_window, int scale_flag)
|
|
{
|
|
Pulse pulse;
|
|
TemporalNoiseShaping *tns = &sce->tns;
|
|
IndividualChannelStream *ics = &sce->ics;
|
|
float *out = sce->coeffs;
|
|
int global_gain, pulse_present = 0;
|
|
|
|
/* This assignment is to silence a GCC warning about the variable being used
|
|
* uninitialized when in fact it always is.
|
|
*/
|
|
pulse.num_pulse = 0;
|
|
|
|
global_gain = get_bits(gb, 8);
|
|
|
|
if (!common_window && !scale_flag) {
|
|
if (decode_ics_info(ac, ics, gb, 0) < 0)
|
|
return -1;
|
|
}
|
|
|
|
if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
|
|
return -1;
|
|
if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
|
|
return -1;
|
|
|
|
pulse_present = 0;
|
|
if (!scale_flag) {
|
|
if ((pulse_present = get_bits1(gb))) {
|
|
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
|
|
return -1;
|
|
}
|
|
if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
|
|
return -1;
|
|
}
|
|
}
|
|
if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
|
|
return -1;
|
|
if (get_bits1(gb)) {
|
|
av_log_missing_feature(ac->avctx, "SSR", 1);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
|
|
return -1;
|
|
|
|
if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
|
|
apply_prediction(ac, sce);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Mid/Side stereo decoding; reference: 4.6.8.1.3.
|
|
*/
|
|
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
|
|
{
|
|
const IndividualChannelStream *ics = &cpe->ch[0].ics;
|
|
float *ch0 = cpe->ch[0].coeffs;
|
|
float *ch1 = cpe->ch[1].coeffs;
|
|
int g, i, group, idx = 0;
|
|
const uint16_t *offsets = ics->swb_offset;
|
|
for (g = 0; g < ics->num_window_groups; g++) {
|
|
for (i = 0; i < ics->max_sfb; i++, idx++) {
|
|
if (cpe->ms_mask[idx] &&
|
|
cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
|
|
for (group = 0; group < ics->group_len[g]; group++) {
|
|
ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
|
|
ch1 + group * 128 + offsets[i],
|
|
offsets[i+1] - offsets[i]);
|
|
}
|
|
}
|
|
}
|
|
ch0 += ics->group_len[g] * 128;
|
|
ch1 += ics->group_len[g] * 128;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* intensity stereo decoding; reference: 4.6.8.2.3
|
|
*
|
|
* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
|
|
* [1] mask is decoded from bitstream; [2] mask is all 1s;
|
|
* [3] reserved for scalable AAC
|
|
*/
|
|
static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
|
|
{
|
|
const IndividualChannelStream *ics = &cpe->ch[1].ics;
|
|
SingleChannelElement *sce1 = &cpe->ch[1];
|
|
float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
|
|
const uint16_t *offsets = ics->swb_offset;
|
|
int g, group, i, idx = 0;
|
|
int c;
|
|
float scale;
|
|
for (g = 0; g < ics->num_window_groups; g++) {
|
|
for (i = 0; i < ics->max_sfb;) {
|
|
if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
|
|
const int bt_run_end = sce1->band_type_run_end[idx];
|
|
for (; i < bt_run_end; i++, idx++) {
|
|
c = -1 + 2 * (sce1->band_type[idx] - 14);
|
|
if (ms_present)
|
|
c *= 1 - 2 * cpe->ms_mask[idx];
|
|
scale = c * sce1->sf[idx];
|
|
for (group = 0; group < ics->group_len[g]; group++)
|
|
ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
|
|
coef0 + group * 128 + offsets[i],
|
|
scale,
|
|
offsets[i + 1] - offsets[i]);
|
|
}
|
|
} else {
|
|
int bt_run_end = sce1->band_type_run_end[idx];
|
|
idx += bt_run_end - i;
|
|
i = bt_run_end;
|
|
}
|
|
}
|
|
coef0 += ics->group_len[g] * 128;
|
|
coef1 += ics->group_len[g] * 128;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Decode a channel_pair_element; reference: table 4.4.
|
|
*
|
|
* @return Returns error status. 0 - OK, !0 - error
|
|
*/
|
|
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
|
|
{
|
|
int i, ret, common_window, ms_present = 0;
|
|
|
|
common_window = get_bits1(gb);
|
|
if (common_window) {
|
|
if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
|
|
return -1;
|
|
i = cpe->ch[1].ics.use_kb_window[0];
|
|
cpe->ch[1].ics = cpe->ch[0].ics;
|
|
cpe->ch[1].ics.use_kb_window[1] = i;
|
|
if (cpe->ch[1].ics.predictor_present && (ac->m4ac.object_type != AOT_AAC_MAIN))
|
|
if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
|
|
decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
|
|
ms_present = get_bits(gb, 2);
|
|
if (ms_present == 3) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
|
|
return -1;
|
|
} else if (ms_present)
|
|
decode_mid_side_stereo(cpe, gb, ms_present);
|
|
}
|
|
if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
|
|
return ret;
|
|
if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
|
|
return ret;
|
|
|
|
if (common_window) {
|
|
if (ms_present)
|
|
apply_mid_side_stereo(ac, cpe);
|
|
if (ac->m4ac.object_type == AOT_AAC_MAIN) {
|
|
apply_prediction(ac, &cpe->ch[0]);
|
|
apply_prediction(ac, &cpe->ch[1]);
|
|
}
|
|
}
|
|
|
|
apply_intensity_stereo(ac, cpe, ms_present);
|
|
return 0;
|
|
}
|
|
|
|
static const float cce_scale[] = {
|
|
1.09050773266525765921, //2^(1/8)
|
|
1.18920711500272106672, //2^(1/4)
|
|
M_SQRT2,
|
|
2,
|
|
};
|
|
|
|
/**
|
|
* Decode coupling_channel_element; reference: table 4.8.
|
|
*
|
|
* @return Returns error status. 0 - OK, !0 - error
|
|
*/
|
|
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
|
|
{
|
|
int num_gain = 0;
|
|
int c, g, sfb, ret;
|
|
int sign;
|
|
float scale;
|
|
SingleChannelElement *sce = &che->ch[0];
|
|
ChannelCoupling *coup = &che->coup;
|
|
|
|
coup->coupling_point = 2 * get_bits1(gb);
|
|
coup->num_coupled = get_bits(gb, 3);
|
|
for (c = 0; c <= coup->num_coupled; c++) {
|
|
num_gain++;
|
|
coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
|
|
coup->id_select[c] = get_bits(gb, 4);
|
|
if (coup->type[c] == TYPE_CPE) {
|
|
coup->ch_select[c] = get_bits(gb, 2);
|
|
if (coup->ch_select[c] == 3)
|
|
num_gain++;
|
|
} else
|
|
coup->ch_select[c] = 2;
|
|
}
|
|
coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
|
|
|
|
sign = get_bits(gb, 1);
|
|
scale = cce_scale[get_bits(gb, 2)];
|
|
|
|
if ((ret = decode_ics(ac, sce, gb, 0, 0)))
|
|
return ret;
|
|
|
|
for (c = 0; c < num_gain; c++) {
|
|
int idx = 0;
|
|
int cge = 1;
|
|
int gain = 0;
|
|
float gain_cache = 1.;
|
|
if (c) {
|
|
cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
|
|
gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
|
|
gain_cache = powf(scale, -gain);
|
|
}
|
|
if (coup->coupling_point == AFTER_IMDCT) {
|
|
coup->gain[c][0] = gain_cache;
|
|
} else {
|
|
for (g = 0; g < sce->ics.num_window_groups; g++) {
|
|
for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
|
|
if (sce->band_type[idx] != ZERO_BT) {
|
|
if (!cge) {
|
|
int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
|
|
if (t) {
|
|
int s = 1;
|
|
t = gain += t;
|
|
if (sign) {
|
|
s -= 2 * (t & 0x1);
|
|
t >>= 1;
|
|
}
|
|
gain_cache = powf(scale, -t) * s;
|
|
}
|
|
}
|
|
coup->gain[c][idx] = gain_cache;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
|
|
*
|
|
* @return Returns number of bytes consumed.
|
|
*/
|
|
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
|
|
GetBitContext *gb)
|
|
{
|
|
int i;
|
|
int num_excl_chan = 0;
|
|
|
|
do {
|
|
for (i = 0; i < 7; i++)
|
|
che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
|
|
} while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
|
|
|
|
return num_excl_chan / 7;
|
|
}
|
|
|
|
/**
|
|
* Decode dynamic range information; reference: table 4.52.
|
|
*
|
|
* @param cnt length of TYPE_FIL syntactic element in bytes
|
|
*
|
|
* @return Returns number of bytes consumed.
|
|
*/
|
|
static int decode_dynamic_range(DynamicRangeControl *che_drc,
|
|
GetBitContext *gb, int cnt)
|
|
{
|
|
int n = 1;
|
|
int drc_num_bands = 1;
|
|
int i;
|
|
|
|
/* pce_tag_present? */
|
|
if (get_bits1(gb)) {
|
|
che_drc->pce_instance_tag = get_bits(gb, 4);
|
|
skip_bits(gb, 4); // tag_reserved_bits
|
|
n++;
|
|
}
|
|
|
|
/* excluded_chns_present? */
|
|
if (get_bits1(gb)) {
|
|
n += decode_drc_channel_exclusions(che_drc, gb);
|
|
}
|
|
|
|
/* drc_bands_present? */
|
|
if (get_bits1(gb)) {
|
|
che_drc->band_incr = get_bits(gb, 4);
|
|
che_drc->interpolation_scheme = get_bits(gb, 4);
|
|
n++;
|
|
drc_num_bands += che_drc->band_incr;
|
|
for (i = 0; i < drc_num_bands; i++) {
|
|
che_drc->band_top[i] = get_bits(gb, 8);
|
|
n++;
|
|
}
|
|
}
|
|
|
|
/* prog_ref_level_present? */
|
|
if (get_bits1(gb)) {
|
|
che_drc->prog_ref_level = get_bits(gb, 7);
|
|
skip_bits1(gb); // prog_ref_level_reserved_bits
|
|
n++;
|
|
}
|
|
|
|
for (i = 0; i < drc_num_bands; i++) {
|
|
che_drc->dyn_rng_sgn[i] = get_bits1(gb);
|
|
che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
|
|
n++;
|
|
}
|
|
|
|
return n;
|
|
}
|
|
|
|
/**
|
|
* Decode extension data (incomplete); reference: table 4.51.
|
|
*
|
|
* @param cnt length of TYPE_FIL syntactic element in bytes
|
|
*
|
|
* @return Returns number of bytes consumed
|
|
*/
|
|
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
|
|
ChannelElement *che, enum RawDataBlockType elem_type)
|
|
{
|
|
int crc_flag = 0;
|
|
int res = cnt;
|
|
switch (get_bits(gb, 4)) { // extension type
|
|
case EXT_SBR_DATA_CRC:
|
|
crc_flag++;
|
|
case EXT_SBR_DATA:
|
|
if (!che) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
|
|
return res;
|
|
} else if (!ac->m4ac.sbr) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
|
|
skip_bits_long(gb, 8 * cnt - 4);
|
|
return res;
|
|
} else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
|
|
skip_bits_long(gb, 8 * cnt - 4);
|
|
return res;
|
|
} else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
|
|
ac->m4ac.sbr = 1;
|
|
ac->m4ac.ps = 1;
|
|
output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
|
|
} else {
|
|
ac->m4ac.sbr = 1;
|
|
}
|
|
res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
|
|
break;
|
|
case EXT_DYNAMIC_RANGE:
|
|
res = decode_dynamic_range(&ac->che_drc, gb, cnt);
|
|
break;
|
|
case EXT_FILL:
|
|
case EXT_FILL_DATA:
|
|
case EXT_DATA_ELEMENT:
|
|
default:
|
|
skip_bits_long(gb, 8 * cnt - 4);
|
|
break;
|
|
};
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
|
|
*
|
|
* @param decode 1 if tool is used normally, 0 if tool is used in LTP.
|
|
* @param coef spectral coefficients
|
|
*/
|
|
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
|
|
IndividualChannelStream *ics, int decode)
|
|
{
|
|
const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
|
|
int w, filt, m, i;
|
|
int bottom, top, order, start, end, size, inc;
|
|
float lpc[TNS_MAX_ORDER];
|
|
float tmp[TNS_MAX_ORDER];
|
|
|
|
for (w = 0; w < ics->num_windows; w++) {
|
|
bottom = ics->num_swb;
|
|
for (filt = 0; filt < tns->n_filt[w]; filt++) {
|
|
top = bottom;
|
|
bottom = FFMAX(0, top - tns->length[w][filt]);
|
|
order = tns->order[w][filt];
|
|
if (order == 0)
|
|
continue;
|
|
|
|
// tns_decode_coef
|
|
compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
|
|
|
|
start = ics->swb_offset[FFMIN(bottom, mmm)];
|
|
end = ics->swb_offset[FFMIN( top, mmm)];
|
|
if ((size = end - start) <= 0)
|
|
continue;
|
|
if (tns->direction[w][filt]) {
|
|
inc = -1;
|
|
start = end - 1;
|
|
} else {
|
|
inc = 1;
|
|
}
|
|
start += w * 128;
|
|
|
|
if (decode) {
|
|
// ar filter
|
|
for (m = 0; m < size; m++, start += inc)
|
|
for (i = 1; i <= FFMIN(m, order); i++)
|
|
coef[start] -= coef[start - i * inc] * lpc[i - 1];
|
|
} else {
|
|
// ma filter
|
|
for (m = 0; m < size; m++, start += inc) {
|
|
tmp[0] = coef[start];
|
|
for (i = 1; i <= FFMIN(m, order); i++)
|
|
coef[start] += tmp[i] * lpc[i - 1];
|
|
for (i = order; i > 0; i--)
|
|
tmp[i] = tmp[i - 1];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Apply windowing and MDCT to obtain the spectral
|
|
* coefficient from the predicted sample by LTP.
|
|
*/
|
|
static void windowing_and_mdct_ltp(AACContext *ac, float *out,
|
|
float *in, IndividualChannelStream *ics)
|
|
{
|
|
const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
|
|
const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
|
|
const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
|
|
const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
|
|
|
|
if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
|
|
ac->dsp.vector_fmul(in, in, lwindow_prev, 1024);
|
|
} else {
|
|
memset(in, 0, 448 * sizeof(float));
|
|
ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
|
|
memcpy(in + 576, in + 576, 448 * sizeof(float));
|
|
}
|
|
if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
|
|
ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
|
|
} else {
|
|
memcpy(in + 1024, in + 1024, 448 * sizeof(float));
|
|
ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
|
|
memset(in + 1024 + 576, 0, 448 * sizeof(float));
|
|
}
|
|
ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
|
|
}
|
|
|
|
/**
|
|
* Apply the long term prediction
|
|
*/
|
|
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
|
|
{
|
|
const LongTermPrediction *ltp = &sce->ics.ltp;
|
|
const uint16_t *offsets = sce->ics.swb_offset;
|
|
int i, sfb;
|
|
|
|
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
|
|
float *predTime = sce->ret;
|
|
float *predFreq = ac->buf_mdct;
|
|
int16_t num_samples = 2048;
|
|
|
|
if (ltp->lag < 1024)
|
|
num_samples = ltp->lag + 1024;
|
|
for (i = 0; i < num_samples; i++)
|
|
predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
|
|
memset(&predTime[i], 0, (2048 - i) * sizeof(float));
|
|
|
|
windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
|
|
|
|
if (sce->tns.present)
|
|
apply_tns(predFreq, &sce->tns, &sce->ics, 0);
|
|
|
|
for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
|
|
if (ltp->used[sfb])
|
|
for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
|
|
sce->coeffs[i] += predFreq[i];
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Update the LTP buffer for next frame
|
|
*/
|
|
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
|
|
{
|
|
IndividualChannelStream *ics = &sce->ics;
|
|
float *saved = sce->saved;
|
|
float *saved_ltp = sce->coeffs;
|
|
const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
|
|
const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
|
|
int i;
|
|
|
|
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
|
|
memcpy(saved_ltp, saved, 512 * sizeof(float));
|
|
memset(saved_ltp + 576, 0, 448 * sizeof(float));
|
|
ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
|
|
for (i = 0; i < 64; i++)
|
|
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
|
|
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
|
|
memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
|
|
memset(saved_ltp + 576, 0, 448 * sizeof(float));
|
|
ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
|
|
for (i = 0; i < 64; i++)
|
|
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
|
|
} else { // LONG_STOP or ONLY_LONG
|
|
ac->dsp.vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
|
|
for (i = 0; i < 512; i++)
|
|
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
|
|
}
|
|
|
|
memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
|
|
memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
|
|
memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
|
|
}
|
|
|
|
/**
|
|
* Conduct IMDCT and windowing.
|
|
*/
|
|
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
|
|
{
|
|
IndividualChannelStream *ics = &sce->ics;
|
|
float *in = sce->coeffs;
|
|
float *out = sce->ret;
|
|
float *saved = sce->saved;
|
|
const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
|
|
const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
|
|
const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
|
|
float *buf = ac->buf_mdct;
|
|
float *temp = ac->temp;
|
|
int i;
|
|
|
|
// imdct
|
|
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
|
|
for (i = 0; i < 1024; i += 128)
|
|
ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
|
|
} else
|
|
ac->mdct.imdct_half(&ac->mdct, buf, in);
|
|
|
|
/* window overlapping
|
|
* NOTE: To simplify the overlapping code, all 'meaningless' short to long
|
|
* and long to short transitions are considered to be short to short
|
|
* transitions. This leaves just two cases (long to long and short to short)
|
|
* with a little special sauce for EIGHT_SHORT_SEQUENCE.
|
|
*/
|
|
if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
|
|
(ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
|
|
ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, 512);
|
|
} else {
|
|
memcpy( out, saved, 448 * sizeof(float));
|
|
|
|
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
|
|
ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
|
|
ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
|
|
ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
|
|
ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
|
|
ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
|
|
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
|
|
} else {
|
|
ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
|
|
memcpy( out + 576, buf + 64, 448 * sizeof(float));
|
|
}
|
|
}
|
|
|
|
// buffer update
|
|
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
|
|
memcpy( saved, temp + 64, 64 * sizeof(float));
|
|
ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
|
|
ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
|
|
ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
|
|
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
|
|
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
|
|
memcpy( saved, buf + 512, 448 * sizeof(float));
|
|
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
|
|
} else { // LONG_STOP or ONLY_LONG
|
|
memcpy( saved, buf + 512, 512 * sizeof(float));
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Apply dependent channel coupling (applied before IMDCT).
|
|
*
|
|
* @param index index into coupling gain array
|
|
*/
|
|
static void apply_dependent_coupling(AACContext *ac,
|
|
SingleChannelElement *target,
|
|
ChannelElement *cce, int index)
|
|
{
|
|
IndividualChannelStream *ics = &cce->ch[0].ics;
|
|
const uint16_t *offsets = ics->swb_offset;
|
|
float *dest = target->coeffs;
|
|
const float *src = cce->ch[0].coeffs;
|
|
int g, i, group, k, idx = 0;
|
|
if (ac->m4ac.object_type == AOT_AAC_LTP) {
|
|
av_log(ac->avctx, AV_LOG_ERROR,
|
|
"Dependent coupling is not supported together with LTP\n");
|
|
return;
|
|
}
|
|
for (g = 0; g < ics->num_window_groups; g++) {
|
|
for (i = 0; i < ics->max_sfb; i++, idx++) {
|
|
if (cce->ch[0].band_type[idx] != ZERO_BT) {
|
|
const float gain = cce->coup.gain[index][idx];
|
|
for (group = 0; group < ics->group_len[g]; group++) {
|
|
for (k = offsets[i]; k < offsets[i + 1]; k++) {
|
|
// XXX dsputil-ize
|
|
dest[group * 128 + k] += gain * src[group * 128 + k];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
dest += ics->group_len[g] * 128;
|
|
src += ics->group_len[g] * 128;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Apply independent channel coupling (applied after IMDCT).
|
|
*
|
|
* @param index index into coupling gain array
|
|
*/
|
|
static void apply_independent_coupling(AACContext *ac,
|
|
SingleChannelElement *target,
|
|
ChannelElement *cce, int index)
|
|
{
|
|
int i;
|
|
const float gain = cce->coup.gain[index][0];
|
|
const float *src = cce->ch[0].ret;
|
|
float *dest = target->ret;
|
|
const int len = 1024 << (ac->m4ac.sbr == 1);
|
|
|
|
for (i = 0; i < len; i++)
|
|
dest[i] += gain * src[i];
|
|
}
|
|
|
|
/**
|
|
* channel coupling transformation interface
|
|
*
|
|
* @param apply_coupling_method pointer to (in)dependent coupling function
|
|
*/
|
|
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
|
|
enum RawDataBlockType type, int elem_id,
|
|
enum CouplingPoint coupling_point,
|
|
void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
|
|
{
|
|
int i, c;
|
|
|
|
for (i = 0; i < MAX_ELEM_ID; i++) {
|
|
ChannelElement *cce = ac->che[TYPE_CCE][i];
|
|
int index = 0;
|
|
|
|
if (cce && cce->coup.coupling_point == coupling_point) {
|
|
ChannelCoupling *coup = &cce->coup;
|
|
|
|
for (c = 0; c <= coup->num_coupled; c++) {
|
|
if (coup->type[c] == type && coup->id_select[c] == elem_id) {
|
|
if (coup->ch_select[c] != 1) {
|
|
apply_coupling_method(ac, &cc->ch[0], cce, index);
|
|
if (coup->ch_select[c] != 0)
|
|
index++;
|
|
}
|
|
if (coup->ch_select[c] != 2)
|
|
apply_coupling_method(ac, &cc->ch[1], cce, index++);
|
|
} else
|
|
index += 1 + (coup->ch_select[c] == 3);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Convert spectral data to float samples, applying all supported tools as appropriate.
|
|
*/
|
|
static void spectral_to_sample(AACContext *ac)
|
|
{
|
|
int i, type;
|
|
for (type = 3; type >= 0; type--) {
|
|
for (i = 0; i < MAX_ELEM_ID; i++) {
|
|
ChannelElement *che = ac->che[type][i];
|
|
if (che) {
|
|
if (type <= TYPE_CPE)
|
|
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
|
|
if (ac->m4ac.object_type == AOT_AAC_LTP) {
|
|
if (che->ch[0].ics.predictor_present) {
|
|
if (che->ch[0].ics.ltp.present)
|
|
apply_ltp(ac, &che->ch[0]);
|
|
if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
|
|
apply_ltp(ac, &che->ch[1]);
|
|
}
|
|
}
|
|
if (che->ch[0].tns.present)
|
|
apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
|
|
if (che->ch[1].tns.present)
|
|
apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
|
|
if (type <= TYPE_CPE)
|
|
apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
|
|
if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
|
|
imdct_and_windowing(ac, &che->ch[0]);
|
|
if (ac->m4ac.object_type == AOT_AAC_LTP)
|
|
update_ltp(ac, &che->ch[0]);
|
|
if (type == TYPE_CPE) {
|
|
imdct_and_windowing(ac, &che->ch[1]);
|
|
if (ac->m4ac.object_type == AOT_AAC_LTP)
|
|
update_ltp(ac, &che->ch[1]);
|
|
}
|
|
if (ac->m4ac.sbr > 0) {
|
|
ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
|
|
}
|
|
}
|
|
if (type <= TYPE_CCE)
|
|
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
|
|
{
|
|
int size;
|
|
AACADTSHeaderInfo hdr_info;
|
|
|
|
size = ff_aac_parse_header(gb, &hdr_info);
|
|
if (size > 0) {
|
|
if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
|
|
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
|
|
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
|
|
ac->m4ac.chan_config = hdr_info.chan_config;
|
|
if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
|
|
return -7;
|
|
if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
|
|
return -7;
|
|
} else if (ac->output_configured != OC_LOCKED) {
|
|
ac->output_configured = OC_NONE;
|
|
}
|
|
if (ac->output_configured != OC_LOCKED) {
|
|
ac->m4ac.sbr = -1;
|
|
ac->m4ac.ps = -1;
|
|
}
|
|
ac->m4ac.sample_rate = hdr_info.sample_rate;
|
|
ac->m4ac.sampling_index = hdr_info.sampling_index;
|
|
ac->m4ac.object_type = hdr_info.object_type;
|
|
if (!ac->avctx->sample_rate)
|
|
ac->avctx->sample_rate = hdr_info.sample_rate;
|
|
if (hdr_info.num_aac_frames == 1) {
|
|
if (!hdr_info.crc_absent)
|
|
skip_bits(gb, 16);
|
|
} else {
|
|
av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
|
|
return -1;
|
|
}
|
|
}
|
|
return size;
|
|
}
|
|
|
|
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
|
|
int *data_size, GetBitContext *gb)
|
|
{
|
|
AACContext *ac = avctx->priv_data;
|
|
ChannelElement *che = NULL, *che_prev = NULL;
|
|
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
|
|
int err, elem_id, data_size_tmp;
|
|
int samples = 0, multiplier;
|
|
|
|
if (show_bits(gb, 12) == 0xfff) {
|
|
if (parse_adts_frame_header(ac, gb) < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
|
|
return -1;
|
|
}
|
|
if (ac->m4ac.sampling_index > 12) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
ac->tags_mapped = 0;
|
|
// parse
|
|
while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
|
|
elem_id = get_bits(gb, 4);
|
|
|
|
if (elem_type < TYPE_DSE) {
|
|
if (!(che=get_che(ac, elem_type, elem_id))) {
|
|
av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
|
|
elem_type, elem_id);
|
|
return -1;
|
|
}
|
|
samples = 1024;
|
|
}
|
|
|
|
switch (elem_type) {
|
|
|
|
case TYPE_SCE:
|
|
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
|
|
break;
|
|
|
|
case TYPE_CPE:
|
|
err = decode_cpe(ac, gb, che);
|
|
break;
|
|
|
|
case TYPE_CCE:
|
|
err = decode_cce(ac, gb, che);
|
|
break;
|
|
|
|
case TYPE_LFE:
|
|
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
|
|
break;
|
|
|
|
case TYPE_DSE:
|
|
err = skip_data_stream_element(ac, gb);
|
|
break;
|
|
|
|
case TYPE_PCE: {
|
|
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
|
|
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
|
|
if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
|
|
break;
|
|
if (ac->output_configured > OC_TRIAL_PCE)
|
|
av_log(avctx, AV_LOG_ERROR,
|
|
"Not evaluating a further program_config_element as this construct is dubious at best.\n");
|
|
else
|
|
err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
|
|
break;
|
|
}
|
|
|
|
case TYPE_FIL:
|
|
if (elem_id == 15)
|
|
elem_id += get_bits(gb, 8) - 1;
|
|
if (get_bits_left(gb) < 8 * elem_id) {
|
|
av_log(avctx, AV_LOG_ERROR, overread_err);
|
|
return -1;
|
|
}
|
|
while (elem_id > 0)
|
|
elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
|
|
err = 0; /* FIXME */
|
|
break;
|
|
|
|
default:
|
|
err = -1; /* should not happen, but keeps compiler happy */
|
|
break;
|
|
}
|
|
|
|
che_prev = che;
|
|
elem_type_prev = elem_type;
|
|
|
|
if (err)
|
|
return err;
|
|
|
|
if (get_bits_left(gb) < 3) {
|
|
av_log(avctx, AV_LOG_ERROR, overread_err);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
spectral_to_sample(ac);
|
|
|
|
multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
|
|
samples <<= multiplier;
|
|
if (ac->output_configured < OC_LOCKED) {
|
|
avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
|
|
avctx->frame_size = samples;
|
|
}
|
|
|
|
data_size_tmp = samples * avctx->channels *
|
|
(av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
|
|
if (*data_size < data_size_tmp) {
|
|
av_log(avctx, AV_LOG_ERROR,
|
|
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
|
|
*data_size, data_size_tmp);
|
|
return -1;
|
|
}
|
|
*data_size = data_size_tmp;
|
|
|
|
if (samples) {
|
|
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
|
|
ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
|
|
samples, avctx->channels);
|
|
else
|
|
ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
|
|
samples, avctx->channels);
|
|
}
|
|
|
|
if (ac->output_configured)
|
|
ac->output_configured = OC_LOCKED;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int aac_decode_frame(AVCodecContext *avctx, void *data,
|
|
int *data_size, AVPacket *avpkt)
|
|
{
|
|
const uint8_t *buf = avpkt->data;
|
|
int buf_size = avpkt->size;
|
|
GetBitContext gb;
|
|
int buf_consumed;
|
|
int buf_offset;
|
|
int err;
|
|
|
|
init_get_bits(&gb, buf, buf_size * 8);
|
|
|
|
if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
|
|
return err;
|
|
|
|
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
|
|
for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
|
|
if (buf[buf_offset])
|
|
break;
|
|
|
|
return buf_size > buf_offset ? buf_consumed : buf_size;
|
|
}
|
|
|
|
static av_cold int aac_decode_close(AVCodecContext *avctx)
|
|
{
|
|
AACContext *ac = avctx->priv_data;
|
|
int i, type;
|
|
|
|
for (i = 0; i < MAX_ELEM_ID; i++) {
|
|
for (type = 0; type < 4; type++) {
|
|
if (ac->che[type][i])
|
|
ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
|
|
av_freep(&ac->che[type][i]);
|
|
}
|
|
}
|
|
|
|
ff_mdct_end(&ac->mdct);
|
|
ff_mdct_end(&ac->mdct_small);
|
|
ff_mdct_end(&ac->mdct_ltp);
|
|
return 0;
|
|
}
|
|
|
|
|
|
#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
|
|
|
|
struct LATMContext {
|
|
AACContext aac_ctx; ///< containing AACContext
|
|
int initialized; ///< initilized after a valid extradata was seen
|
|
|
|
// parser data
|
|
int audio_mux_version_A; ///< LATM syntax version
|
|
int frame_length_type; ///< 0/1 variable/fixed frame length
|
|
int frame_length; ///< frame length for fixed frame length
|
|
};
|
|
|
|
static inline uint32_t latm_get_value(GetBitContext *b)
|
|
{
|
|
int length = get_bits(b, 2);
|
|
|
|
return get_bits_long(b, (length+1)*8);
|
|
}
|
|
|
|
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
|
|
GetBitContext *gb)
|
|
{
|
|
AVCodecContext *avctx = latmctx->aac_ctx.avctx;
|
|
MPEG4AudioConfig m4ac;
|
|
int config_start_bit = get_bits_count(gb);
|
|
int bits_consumed, esize;
|
|
|
|
if (config_start_bit % 8) {
|
|
av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
|
|
"config not byte aligned.\n", 1);
|
|
return AVERROR_INVALIDDATA;
|
|
} else {
|
|
bits_consumed =
|
|
decode_audio_specific_config(NULL, avctx, &m4ac,
|
|
gb->buffer + (config_start_bit / 8),
|
|
get_bits_left(gb) / 8);
|
|
|
|
if (bits_consumed < 0)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
esize = (bits_consumed+7) / 8;
|
|
|
|
if (avctx->extradata_size <= esize) {
|
|
av_free(avctx->extradata);
|
|
avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
|
|
if (!avctx->extradata)
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
avctx->extradata_size = esize;
|
|
memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
|
|
memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
|
|
|
|
skip_bits_long(gb, bits_consumed);
|
|
}
|
|
|
|
return bits_consumed;
|
|
}
|
|
|
|
static int read_stream_mux_config(struct LATMContext *latmctx,
|
|
GetBitContext *gb)
|
|
{
|
|
int ret, audio_mux_version = get_bits(gb, 1);
|
|
|
|
latmctx->audio_mux_version_A = 0;
|
|
if (audio_mux_version)
|
|
latmctx->audio_mux_version_A = get_bits(gb, 1);
|
|
|
|
if (!latmctx->audio_mux_version_A) {
|
|
|
|
if (audio_mux_version)
|
|
latm_get_value(gb); // taraFullness
|
|
|
|
skip_bits(gb, 1); // allStreamSameTimeFraming
|
|
skip_bits(gb, 6); // numSubFrames
|
|
// numPrograms
|
|
if (get_bits(gb, 4)) { // numPrograms
|
|
av_log_missing_feature(latmctx->aac_ctx.avctx,
|
|
"multiple programs are not supported\n", 1);
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
|
|
// for each program (which there is only on in DVB)
|
|
|
|
// for each layer (which there is only on in DVB)
|
|
if (get_bits(gb, 3)) { // numLayer
|
|
av_log_missing_feature(latmctx->aac_ctx.avctx,
|
|
"multiple layers are not supported\n", 1);
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
|
|
// for all but first stream: use_same_config = get_bits(gb, 1);
|
|
if (!audio_mux_version) {
|
|
if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
|
|
return ret;
|
|
} else {
|
|
int ascLen = latm_get_value(gb);
|
|
if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
|
|
return ret;
|
|
ascLen -= ret;
|
|
skip_bits_long(gb, ascLen);
|
|
}
|
|
|
|
latmctx->frame_length_type = get_bits(gb, 3);
|
|
switch (latmctx->frame_length_type) {
|
|
case 0:
|
|
skip_bits(gb, 8); // latmBufferFullness
|
|
break;
|
|
case 1:
|
|
latmctx->frame_length = get_bits(gb, 9);
|
|
break;
|
|
case 3:
|
|
case 4:
|
|
case 5:
|
|
skip_bits(gb, 6); // CELP frame length table index
|
|
break;
|
|
case 6:
|
|
case 7:
|
|
skip_bits(gb, 1); // HVXC frame length table index
|
|
break;
|
|
}
|
|
|
|
if (get_bits(gb, 1)) { // other data
|
|
if (audio_mux_version) {
|
|
latm_get_value(gb); // other_data_bits
|
|
} else {
|
|
int esc;
|
|
do {
|
|
esc = get_bits(gb, 1);
|
|
skip_bits(gb, 8);
|
|
} while (esc);
|
|
}
|
|
}
|
|
|
|
if (get_bits(gb, 1)) // crc present
|
|
skip_bits(gb, 8); // config_crc
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
|
|
{
|
|
uint8_t tmp;
|
|
|
|
if (ctx->frame_length_type == 0) {
|
|
int mux_slot_length = 0;
|
|
do {
|
|
tmp = get_bits(gb, 8);
|
|
mux_slot_length += tmp;
|
|
} while (tmp == 255);
|
|
return mux_slot_length;
|
|
} else if (ctx->frame_length_type == 1) {
|
|
return ctx->frame_length;
|
|
} else if (ctx->frame_length_type == 3 ||
|
|
ctx->frame_length_type == 5 ||
|
|
ctx->frame_length_type == 7) {
|
|
skip_bits(gb, 2); // mux_slot_length_coded
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int read_audio_mux_element(struct LATMContext *latmctx,
|
|
GetBitContext *gb)
|
|
{
|
|
int err;
|
|
uint8_t use_same_mux = get_bits(gb, 1);
|
|
if (!use_same_mux) {
|
|
if ((err = read_stream_mux_config(latmctx, gb)) < 0)
|
|
return err;
|
|
} else if (!latmctx->aac_ctx.avctx->extradata) {
|
|
av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
|
|
"no decoder config found\n");
|
|
return AVERROR(EAGAIN);
|
|
}
|
|
if (latmctx->audio_mux_version_A == 0) {
|
|
int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
|
|
if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
|
|
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
|
|
return AVERROR_INVALIDDATA;
|
|
} else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
|
|
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
|
|
"frame length mismatch %d << %d\n",
|
|
mux_slot_length_bytes * 8, get_bits_left(gb));
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
|
|
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
|
|
AVPacket *avpkt)
|
|
{
|
|
struct LATMContext *latmctx = avctx->priv_data;
|
|
int muxlength, err;
|
|
GetBitContext gb;
|
|
|
|
if (avpkt->size == 0)
|
|
return 0;
|
|
|
|
init_get_bits(&gb, avpkt->data, avpkt->size * 8);
|
|
|
|
// check for LOAS sync word
|
|
if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
muxlength = get_bits(&gb, 13) + 3;
|
|
// not enough data, the parser should have sorted this
|
|
if (muxlength > avpkt->size)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
|
|
return err;
|
|
|
|
if (!latmctx->initialized) {
|
|
if (!avctx->extradata) {
|
|
*out_size = 0;
|
|
return avpkt->size;
|
|
} else {
|
|
if ((err = aac_decode_init(avctx)) < 0)
|
|
return err;
|
|
latmctx->initialized = 1;
|
|
}
|
|
}
|
|
|
|
if (show_bits(&gb, 12) == 0xfff) {
|
|
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
|
|
"ADTS header detected, probably as result of configuration "
|
|
"misparsing\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
|
|
return err;
|
|
|
|
return muxlength;
|
|
}
|
|
|
|
av_cold static int latm_decode_init(AVCodecContext *avctx)
|
|
{
|
|
struct LATMContext *latmctx = avctx->priv_data;
|
|
int ret;
|
|
|
|
ret = aac_decode_init(avctx);
|
|
|
|
if (avctx->extradata_size > 0) {
|
|
latmctx->initialized = !ret;
|
|
} else {
|
|
latmctx->initialized = 0;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
|
|
AVCodec ff_aac_decoder = {
|
|
"aac",
|
|
AVMEDIA_TYPE_AUDIO,
|
|
CODEC_ID_AAC,
|
|
sizeof(AACContext),
|
|
aac_decode_init,
|
|
NULL,
|
|
aac_decode_close,
|
|
aac_decode_frame,
|
|
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
|
|
.sample_fmts = (const enum AVSampleFormat[]) {
|
|
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
|
|
},
|
|
.channel_layouts = aac_channel_layout,
|
|
};
|
|
|
|
/*
|
|
Note: This decoder filter is intended to decode LATM streams transferred
|
|
in MPEG transport streams which only contain one program.
|
|
To do a more complex LATM demuxing a separate LATM demuxer should be used.
|
|
*/
|
|
AVCodec ff_aac_latm_decoder = {
|
|
.name = "aac_latm",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = CODEC_ID_AAC_LATM,
|
|
.priv_data_size = sizeof(struct LATMContext),
|
|
.init = latm_decode_init,
|
|
.close = aac_decode_close,
|
|
.decode = latm_decode_frame,
|
|
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
|
|
.sample_fmts = (const enum AVSampleFormat[]) {
|
|
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
|
|
},
|
|
.channel_layouts = aac_channel_layout,
|
|
};
|